> Message: 10 > Date: Sun, 15 May 2005 21:41:23 +0000 > From: Laurent Lesage <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] skype channel > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi *, > > I was just going to ask the same question. Does anybody have an > information about Skype and Asterisk? Any link? > > Thanks in advance
I've just added a view day's ago some information on it on the wiki. As far as I know there is nothing really working 'yet' but I'm sure since the API is out it' won't take long :-) http://www.voip-info.org/tiki-index.php?page=bounty%20skype Wessel de Roode > > Laurent > > > Bartek Kania a icrit : > > >-----BEGIN PGP SIGNED MESSAGE----- > >Hash: SHA1 > > > >I just noticed that the Skype API for linux seems to be available. > >I've read before a number of posts where people were talking about > >implementing a chan_skype with the skype API. > > > >I wonder if there is any progress in that direction, and if anyone is > >working on it. > > > >/B > >- -- > >* GPG-Key: http://evil.gnarf.org/mrbk.pgp > > > >A: Because we read from top to bottom, left to right. > >Q: Why should i start my reply below the quoted text? > >- -- http://www.i-hate-computers.demon.co.uk/ > > > >-----BEGIN PGP SIGNATURE----- > >Version: GnuPG v1.2.5 (GNU/Linux) > > > >iD8DBQFCgLlVWYjaxM2wIe4RAuSKAJ9VNMIO2h838Y2yXAFDAQaJOjPa3gCfeokZ > >Ghsrpa8Gp3pHt5/bUinZKUA= > >=fUgt > >-----END PGP SIGNATURE----- > >_______________________________________________ > >Asterisk-Users mailing list > >Asterisk-Users@lists.digium.com > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > ------------------------------ > > Message: 11 > Date: Sun, 15 May 2005 17:49:41 -0400 > From: Paul <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] knopsterisk > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > trixter http://www.0xdecafbad.com wrote: > > >does anyone have knopsterisk for download, I assume that > because its GPL > >the creator of that iso cant restrict spreading it. A > friend wanted it > >to play on a box and the only thing I can find with google is the > >knopsterisk.com site which wants $10 to get a copy and does > not provide > >(as far as I can tell) any free distribution access which is > >his/hers/its/them/they/whatever right (being politically correct is > >hard). > > > >If there is some distribution problem with doing this then I > would also > >appreciate hearing why it cant be distroed by 3rd parties. > > > >Thanks > > > > > The website says "/*Now with Asterisk Version 1.0!" which makes me > wonder how many they have sold. Also makes me wonder if the > knoppix part > is very up to date. > > They don't mention licensing/copyright anywhere. We can > figure that all > the software is on the CD is under free licensing but all > they have to > do is add a single readme file with a restricted license or copyright > and you make identical copies of the CD. I would first try contacting > them and get those details. You also want to know where the source is > because there might be some modifications they made to > knoppix packages > or the packages they added. > > I think you would be better off to make a knoppix CD, boot it > and get * > installed and running. After that read the following and > maybe you can > create something better to share with the world. > > http://www.knoppix.net/wiki/Knoppix_Remastering_Howto > > */ > > > ------------------------------ > > Message: 12 > Date: Sun, 15 May 2005 15:55:03 -0600 > From: Ira Burton <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] 911 Options > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > I am curious if anybody has pointers on the best way to get the 7 > digit PSAP number for an area. I am thinking about making a '911' > extension that will dial the PSAP number, wait for the PSAP to answer > and play a message giving the address of the originating call, and > replay the the information every three minutes. I am concerned what > may happen if my children try to dial 911 in an emergency but do not > yet know our address. > > How are other people handling this? > > > ------------------------------ > > Message: 13 > Date: Sun, 15 May 2005 15:15:15 -0700 > From: "trixter http://www.0xdecafbad.com" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] knopsterisk > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > On Sun, 2005-05-15 at 17:49 -0400, Paul wrote: > > I think you would be better off to make a knoppix CD, boot > it and get * > > installed and running. After that read the following and > maybe you can > > create something better to share with the world. > > Unfortunately that wont help my friend who wanted to play but not > reformat his disk. I already have a debian system running > asterisk, so > I personally wouldnt get anything (other than having to redo all my > configs :P > > I just didnt know if it was restricted in any way which would prevent > someone else who had a copy from making it available, and if > someone on > here does have a copy if they could make it available, in the > hopes that > maybe today my friend could download it. > > > -- > Trixter http://www.0xdecafbad.com > UK +44 870 340 4605 Germany +49 801 777 555 3402 > US +1 360 207 0479 or +1 516 687 5200 > FreeWorldDialup: 635378 > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/2 > 0050515/8407774b/attachment-0001.pgp > > ------------------------------ > > Message: 14 > Date: Sun, 15 May 2005 15:17:53 -0700 > From: "trixter http://www.0xdecafbad.com" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] 911 Options > To: Ira Burton <[EMAIL PROTECTED]>, Asterisk Users > Mailing List - > Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > On Sun, 2005-05-15 at 15:55 -0600, Ira Burton wrote: > > I am curious if anybody has pointers on the best way to get the 7 > > digit PSAP number for an area. I am thinking about making a '911' > > extension that will dial the PSAP number, wait for the PSAP > to answer > > and play a message giving the address of the originating call, and > > replay the the information every three minutes. I am concerned what > > may happen if my children try to dial 911 in an emergency but do not > > yet know our address. > > > > You can buy them on CD, however to do E911 you have to have a special > trunk to the switch that the PSAP is off of, which transmits > the E parts > of E911 not just the audio. > > Where to buy them I dont know offhand, I do specifically recall seeing > pages that sold national CDs (how adt, onstar, even other > PSAPs contact > a specific PSAP when needed). > > I do remember that I was googling psap administrators and other such > things. > > > -- > Trixter http://www.0xdecafbad.com > UK +44 870 340 4605 Germany +49 801 777 555 3402 > US +1 360 207 0479 or +1 516 687 5200 > FreeWorldDialup: 635378 > -------------- next part -------------- > A non-text attachment was scrubbed... > Name: not available > Type: application/pgp-signature > Size: 189 bytes > Desc: This is a digitally signed message part > Url : > http://lists.digium.com/pipermail/asterisk-users/attachments/2 > 0050515/713468c7/attachment-0001.pgp > > ------------------------------ > > Message: 15 > Date: Mon, 16 May 2005 07:20:47 +0900 (JST) > From: Zen Kato <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] can't CLI> STOP NOW by zombie MOH > To: asterisk-users@lists.digium.com > Message-ID: <[EMAIL PROTECTED]> > Content-Type: Text/Plain; charset=us-ascii > > I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH > fine. After I stop MOH on Windows Messenger, if the hungup > signal could > not send to *, the sip channel(e.g.,SIP/52001-08ca) for this > MOH remains. > Then the user trys again MOH, a new sip channel starts. And again > the hugup signal can not send to *,......... > > When I 'stop now' from CLI> , * cleanups the remaining sip > channels as follows; > > *CLI> stop now > Beginning asterisk shutdown.... > -- Stopped music on hold on SIP/52001-9e3b > == Spawn extension (sip, 6000, 2) exited non-zero on > 'SIP/52001-9e3b' > -- Stopped music on hold on SIP/52001-08ca > -- Stopped music on hold on SIP/52001-63fd > == Spawn extension (sip, 6000, 2) exited non-zero on > 'SIP/52001-63fd' > == Spawn extension (sip, 6000, 2) exited non-zero on > 'SIP/52001-08ca' > -- Executing Hangup("SIP/52001-9e3b", "") in new stack > == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52001-9e3b' > -- Executing Hangup("SIP/52001-08ca", "") in new stack > == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52001-08ca' > -- Executing Hangup("SIP/52001-63fd", "") in new stack > == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52001-63fd' > Executing last minute cleanups > == Destroying any remaining musiconhold processes > > The CPU goes to 99% usage, but can't stop mpg123 and asterisk, > the following 'top' show ; > > 5737 root 25 0 3832 2588 2424 R 99.7 0.5 2:54.00 mpg123 > 5397 root 15 0 167m 18m 6212 S 2.7 3.7 0:10.95 X > 5627 root 15 0 47988 6604 3624 S 1.3 1.3 0:15.64 asterisk > 4673 root 17 0 8588 6820 1620 S 0.3 1.3 0:02.05 hald > 5509 zenkato 15 0 24168 9.8m 7068 S 0.3 2.0 0:00.47 metacity > 5551 zenkato 15 0 58500 17m 10m S 0.3 3.4 0:01.20 > gnome-terminal > ....(snip)... > > So, I have to do 'kill -9 pid-of-mpg123'. * goes to > segmentation fault. > > Is this safe way to stop asterisk? > > When UA could not send 'hangup signal' to *, the following > warnig came out > console. > > -- Executing Answer("SIP/52003-f48d", "") in new stack > -- Executing MusicOnHold("SIP/52003-f48d", "") in new stack > -- Started music on hold, class 'default', on SIP/52003-f48d > -- Stopped music on hold on SIP/52003-f48d > == Spawn extension (sip, 6000, 2) exited non-zero on > 'SIP/52003-f48d' > -- Executing Hangup("SIP/52003-f48d", "") in new stack > == Spawn extension (sip, h, 1) exited non-zero on 'SIP/52003-f48d' > -- Registered SIP '52001' at 192.168.0.12 port 9558 expires 120 > -- Saved useragent "RTC/1.3.5369 (Messenger 5.1.0639)" > for peer 52001 > -- Executing Answer("SIP/52001-9e3b", "") in new stack > -- Executing MusicOnHold("SIP/52001-9e3b", "") in new stack > -- Started music on hold, class 'default', on SIP/52001-9e3b > May 16 05:45:50 WARNING[5627]: chan_sip.c:695 retrans_pkt: > Maximum retries exceeded on call > a4ea6066f3e34d5aa5ce56b09ea520bf for seqno 1 (Non-critical > Response)Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > May 16 06:01:25 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:01:45 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:02:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > ........(snip)..... > May 16 06:12:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > Warning, flexibel rate not heavily tested! > May 16 06:12:35 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:12:55 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > -- Executing Answer("SIP/52001-63fd", "") in new stack > -- Executing MusicOnHold("SIP/52001-63fd", "") in new stack > -- Started music on hold, class 'default', on SIP/52001-63fd > May 16 06:13:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:13:09 WARNING[5627]: chan_sip.c:695 retrans_pkt: > Maximum retries exceeded on call > 0dc3a12940a34f21bd7f816ef45d0f75 for seqno 1 (Non-critical > Response)May 16 06:13:25 NOTICE[5627]: rtp.c:355 > ast_rtcp_read: RTP: Received packet with bad UDP checksum > -- Executing Answer("SIP/52001-08ca", "") in new stack > -- Executing MusicOnHold("SIP/52001-08ca", "") in new stack > -- Started music on hold, class 'default', on SIP/52001-08ca > May 16 06:13:39 WARNING[5627]: chan_sip.c:695 retrans_pkt: > Maximum retries exceeded on call > 4e503aa4498b45e89632b34034b5e9f1 for seqno 1 (Non-critical > Response)May 16 06:13:45 NOTICE[5627]: rtp.c:355 > ast_rtcp_read: RTP: Received packet with bad UDP checksum > May 16 06:13:54 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:14:14 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:14:14 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > May 16 06:14:15 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: > Received packet with bad UDP checksum > > > Regards, > > Zen Kato > > > > > ------------------------------ > > Message: 16 > Date: Sun, 15 May 2005 18:35:27 -0400 > From: "Chris Mason (Lists)" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Road Warrior phone config > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > > You serious? I typed all that and you were asking about > > RETRIEVING vm all along? Wow, I must be really dense today. > > No, I know it now :-) > > > So: don't pass calleridnum to extension 8500. Or configure a > > different voicemail retrieval exten for roaming users and > > pass null to voicemailmain. > > Yes, that could work. > > > Or, even better, scrape off and discard the fourth extension > > digit when parsing calleridnum and handing to voicemailmain. > > I like that better. > > Thanks > > Chris Mason > www.anguillaguide.com > Tel: (305) 704-7249 Fax: (815)301-9759 > > > > ------------------------------ > > Message: 17 > Date: Sun, 15 May 2005 16:47:33 -0600 > From: Andres Paglayan <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Road Warrior phone config > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1 > > question about this thread, > would a wi-fi voip phone work for this guy? > meaning, he takes it to wherever he goes and it gets > registered wherever > it as wireless access. > is that theoretically correct? > > > > > > > > > > > ------------------------------ > > Message: 18 > Date: Mon, 16 May 2005 00:58:25 +0200 > From: "Thierry Wehr" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Compile problem on last CVS > To: <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Good evening > > from the CVS of the 2005/05/14 it's impossible to build asterisk* on a > redhat 7.3 > > i get this at compile time > > chan_sip.c: In function `build_user': > chan_sip.c:10007: parse error before `struct' > chan_sip.c:10029: `userflags' undeclared (first use in this function) > chan_sip.c:10029: (Each undeclared identifier is reported only once > chan_sip.c:10029: for each function it appears in.) > chan_sip.c:10029: `mask' undeclared (first use in this function) > chan_sip.c:10094: warning: type defaults to `int' in > declaration of `__s' > chan_sip.c:10094: warning: comparison of distinct pointer > types lacks a cast > chan_sip.c: In function `build_peer': > chan_sip.c:10176: parse error before `struct' > chan_sip.c:10221: `peerflags' undeclared (first use in this function) > chan_sip.c:10221: `mask' undeclared (first use in this function) > chan_sip.c:10391: warning: type defaults to `int' in > declaration of `__s' > chan_sip.c:10391: warning: comparison of distinct pointer > types lacks a cast > make[1]: *** [chan_sip.o] Erreur 1 > make[1]: Quitte le ripertoire > `/usr/src/asterisk-cvs/asterisk/channels' > make: *** [subdirs] Erreur 1 > > may be someone have a clue to fix it > > best rehards > Thierry > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/2 > 0050516/0b5498e9/attachment-0001.htm > > ------------------------------ > > Message: 19 > Date: Sun, 15 May 2005 18:03:50 -0500 > From: "Tim Connolly" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Compile problem on last CVS > To: <[EMAIL PROTECTED]>, "'Asterisk Users Mailing List - Non-Commercial > Discussion'" <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Maybe try a version of redhat that was released in the past 5 years? > Seriously, why do you require RH7.3 over Fedora or even RH 9? > > > > _____ > > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Thierry Wehr > Sent: Sunday, May 15, 2005 5:58 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Compile problem on last CVS > > > > Good evening > > > > from the CVS of the 2005/05/14 it's impossible to build asterisk* on a > redhat 7.3 > > > > i get this at compile time > > > > chan_sip.c: In function `build_user': > chan_sip.c:10007: parse error before `struct' > chan_sip.c:10029: `userflags' undeclared (first use in this function) > chan_sip.c:10029: (Each undeclared identifier is reported only once > chan_sip.c:10029: for each function it appears in.) > chan_sip.c:10029: `mask' undeclared (first use in this function) > chan_sip.c:10094: warning: type defaults to `int' in > declaration of `__s' > chan_sip.c:10094: warning: comparison of distinct pointer > types lacks a cast > chan_sip.c: In function `build_peer': > chan_sip.c:10176: parse error before `struct' > chan_sip.c:10221: `peerflags' undeclared (first use in this function) > chan_sip.c:10221: `mask' undeclared (first use in this function) > chan_sip.c:10391: warning: type defaults to `int' in > declaration of `__s' > chan_sip.c:10391: warning: comparison of distinct pointer > types lacks a cast > make[1]: *** [chan_sip.o] Erreur 1 > make[1]: Quitte le ripertoire > `/usr/src/asterisk-cvs/asterisk/channels' > make: *** [subdirs] Erreur 1 > > > > may be someone have a clue to fix it > > > > best rehards > > Thierry > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/2 > 0050515/68f6d234/attachment-0001.htm > > ------------------------------ > > Message: 20 > Date: Sun, 15 May 2005 19:29:58 -0400 > From: Paul <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Road Warrior phone config > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Andres Paglayan wrote: > > >question about this thread, > >would a wi-fi voip phone work for this guy? > >meaning, he takes it to wherever he goes and it gets > registered wherever > >it as wireless access. > >is that theoretically correct? > > > > > I like that approach. Those toys will be getting more affordable. One > concern I would have is battery life. I think a wisip phone > that can be > recharged/powered via standard usb cable would be nice. > > > > ------------------------------ > > Message: 21 > Date: Mon, 16 May 2005 00:33:02 +0100 > From: Tony Hoyle <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] knopsterisk > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > trixter http://www.0xdecafbad.com wrote: > > does anyone have knopsterisk for download, I assume that > because its GPL > > the creator of that iso cant restrict spreading it. A > friend wanted it > > to play on a box and the only thing I can find with google is the > > knopsterisk.com site which wants $10 to get a copy and does > not provide > > (as far as I can tell) any free distribution access which is > > his/hers/its/them/they/whatever right (being politically correct is > > hard). > > The GPL does allow the creator to charge a redistribution > charge... it > doesn't require free distribution (which would be a bit harsh > for some > projects - bandwidth isn't free, and CDs/Burners certainly aren't). > > The CD may contain items of proprietary software - until > recently SuSE > was like this, and Redhat RHEL still is. In that case you can > redistribute most of the contents of the CD but not the CD > itself (ie. > not a working copy, especially if the proprietary part is the > installer). In that case your option is to strip the GPL parts and > build a new distribution, wait for someone to do the same, or > pay for a > copy. > > If it doesn't contain proprietary parts, and all its contents > are freely > licensed, just find someone who's paid for a copy and dupicate > it/download it from them. > > Tony > > > ------------------------------ > > Message: 22 > Date: Sun, 15 May 2005 18:33:37 -0500 > From: "Tim Connolly" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] Road Warrior phone config > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Or have a small solar panel on the back of the phone. Stick > it on the dash > of your car, assuming it doesn't burst into flames from heat; > it should be > fully charged in an hour or two. > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Paul > Sent: Sunday, May 15, 2005 6:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Road Warrior phone config > > Andres Paglayan wrote: > > >question about this thread, > >would a wi-fi voip phone work for this guy? > >meaning, he takes it to wherever he goes and it gets > registered wherever > >it as wireless access. > >is that theoretically correct? > > > > > I like that approach. Those toys will be getting more affordable. One > concern I would have is battery life. I think a wisip phone > that can be > recharged/powered via standard usb cable would be nice. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 23 > Date: Sun, 15 May 2005 18:53:19 -0500 > From: "Tim Connolly" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] FXO/FXS suggestions: > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > I'm looking for a zaptel type device with one (or > more) FXO and > one (or more) FXS port. Basically this guy would sit in-line > of your phone > line (PCI card). Any suggestions? TDM400 would be overkill. > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/2 > 0050515/d66e6335/attachment.htm > > ------------------------------ > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 10, Issue 117 > *********************************************** > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 10-05-05 > > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.11.8 - Release Date: 10-05-05 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users